The field of the invention is that of signal processing, more specifically in the context of correcting distortion produced by an amplifier system. Correction involves comparing the characteristics of the output signal with the original signal at the input of the amplifier system in order to determine the distortion that the system produces. This makes it possible to respond by calculating inverse predistortion to be applied to the input signal to cancel the distortion and thereby improve the linearity and the electrical efficiency of the amplifier system.
The invention is based on the observation that, because the amplifier system introduces a time-delay, the comparison is not applied to the signal and to its exact replica (ignoring the intrinsic distortion of the system), and that this leads to non-negligible error and puts a theoretical limit on how well predistortion can be calculated.
Existing predistortion techniques do not address this problem of error associated with the time-delay between the signals compared.
Because the signals are generally compared using digital techniques, it is necessary to digitize at least the analog output signal of the amplifier system. The digitized output signal at a given time t0 is the replica, distorted to a greater or lesser degree by the amplifier, of the signal on the input side of the system at time t0-xcex4, where xcex4 is the time-delay caused by the system.
Note further that the time-delay is not fixed, and can vary with aging of the system, its temperature, the amplified signal, the power, the frequency band used, etc., which makes correction by applying a single fixed time shift to the comparison signal (reference signal) from the input side of the amplifier system somewhat ineffective.
FIG. 1 is a simplified diagram of the main components of a conventional amplifier system for a radio transmitter in which nonlinearity is corrected by predistortion. The system 2 includes a digital predistortion unit 4 which receives a digital signal X to be amplified at a first input 4a, in this instance from a digital modulation source 6. The unit 4 adds predistortion to the digital signal X, thereby modifying it to produce a predistorted signal Xxe2x80x2 at the unit""s output 4b. The signal Xxe2x80x2 is then converted into analog form by a digital-to-analog converter (DAC) 8 whose output feeds a transmit frequency converter stage (TX-IF and TX-RF) 10 which drives the power amplifier 12, which is connected to a transmit antenna (not shown).
To make the comparison, the output signal Y of the power amplifier 12 is fed back via a loop 14 to a second input 4c of the predistortion unit 4. The loop includes a radio frequency (RF) head 16 whose input receives the output signal of the power amplifier 12 and supplies it in a suitable form to a receiver frequency converter stage (XR-IF and XR-RF) 18 whose function is the inverse of that of the converter stage 10 (even if the intermediate frequencies are not the same). The output of the converter stage 18 is digitized by an analog-to-digital converter (ADC) 20 which supplies the digitized signal Y to the second input 4c of the predistortion unit.
The DAC 8 and the ADC 20 are clocked by a signal "PHgr" spl at respective clock inputs E"PHgr". The signal "PHgr" spl fixes the period of each conversion of the instantaneous digital value of the signal X for the DAC and the period of each digitization of the instantaneous analog value of the signal detected at the output of the power amplifier 12, i.e. the sampling rate. In this example, the signal "PHgr" spl is supplied in common to the DAC 8 and the ADC 20 by a phase-locked loop (PLL) 21 fed by a reference clock "PHgr" ref.
Note that the signals X and Y compared by the predistortion unit are subject to varying time shifts which can exceed the period of the clock "PHgr" spl and whose exact values do not correspond to an integer number of periods of that clock.
In the prior art, the base band output Y from the ADC 20 is compared directly to the signal X, with no further precautions.
This results in limitations on performance caused both by the intrinsic calibration in the factory of phase alignment and most importantly of sampling time alignment between sending (sampling in the DAC 8) and receiving (sampling in the ADC 20), and also by inherent propagation times in the system, which are non-negligible and liable to vary, in particular because of the presence of analog filters. Although the time shifts for the digital signals can be controlled, the problem is much more difficult in the analog domain.
This defective time alignment is particularly prejudicial to obtaining good predistortion performance at reasonable cost in modern transmitter stations, such as those used in third generation cellular systems. These systems use code division multiple access (CDMA) coding at the radio interface. For reasons of cost and size, the base transceiver stations (BTS) use only one transmit radio power amplifier 12 to transmit the signals of all users on one or more carriers. In this context, the term xe2x80x9ctransmissionxe2x80x9d refers to the complete transmission system in the base transceiver station, which includes the digital information and signal processing part and the purely analog part including in particular the power amplifier; the term xe2x80x9cradioxe2x80x9d refers to the part where radio emission proper takes place which comprises the last link of the chain.
The error in the predistortion calculated by this time shift could be corrected using algorithms, but this would require expensive calculation resources and would be based on approximations and extrapolations that would impose a fundamental limitation on accuracy.
Optimum use of new predistortion techniques allied with peak limiting techniques can achieve power amplifier efficiencies up to 15% to 17%.
This represents a significant advance over conventional amplification techniques known as xe2x80x9cfeedforwardxe2x80x9d techniques, the efficiency of which is limited to around 8%. Also, it reduces costs and substitutes digital processing for the complex techniques used in the high power radio frequency and analog stages.
However, to accommodate the many different types of power amplifier and transistor available, and to minimize, or even eliminate, calibration in the factory, the predistortion system must be adaptive so that it can act dynamically to track and correct changes in the transfer function of a power amplifier, especially variations in nonlinearity over the frequency band, in the number of carriers, in temperature, in aging, in memory effects (remanence), etc.
The best and fastest methods are based on conversion tables which construct the inverse of the nonlinearity of a power amplifier and apply it to the modulation signal to reproduce at the power amplifier output the required original signal with the best spectrum masking on adjacent channels.
However, and whichever variant is employed, the algorithms used to update the predistortion table are based on very linear wideband receivers producing a good copy of the radio signal as transmitted. Updating the predistortion table is based on comparing the incoming signal and the outgoing signal, that is to say on the correlation between the incoming signal and the outgoing signal. The greater the radio transmit band and the greater the band to be linearized (at least three to five times the instantaneous transmission band), the more directly is linearization gain linked to the accuracy of the comparison. Most of these methods, based on least mean squares (LMS) algorithms, are relatively insensitive to the quadrature and the gain of the signals to be compared. However, the major problem is the accuracy of the time-delays between the signals compared. In fact, the problem is in all regards identical to that of wideband time-delay locking for a xe2x80x9cfeedforwardxe2x80x9d amplifier, except that it occurs in the digital domain in the case of predistortion. In this case, the difficulty of the problem increases as the width of the band to be linearized increases. In practice, to obtain an improvement in linearity of the order of 25 decibels (dB) for an UMTS amplifier for three or four carriers in a 60 megahertz (MHz) band, the time shift between the transmitted and measured signal samples must not exceed 10 picoseconds. Because of even greater variations in clock phase drift and variations throughout the transmit and receive system, this accuracy cannot be obtained by calibration in the factory.
The problem to be solved is that of finding methods for tracking and measuring time alignment dynamically and for correcting it using simple means that are preferably based on low-cost digital techniques.
A high-efficiency transmitter is used in the prior art, with peak limiting and digital predistortion in tandem. The best-known digital predistortion techniques are based on time control, i.e. a highly linear wideband receiver, and preferably use undersampling at the intermediate frequency (to prevent DC component and quadrature and gain pairing defects) to feed LMS algorithms for updating the predistortion table.
Disturbance of time alignment by signal level, gain or quadrature (in the complex plane) mismatches between the two signals to be compared is prevented. The proposed methods are not directly sensitive to the gain difference between the signals X and Y (see FIG. 1). The same applies to quadrature defects.
LMS algorithms are able to take account of phase rotation and the gain difference between the measured signal and the signal to be transmitted. This applies a constant complex value to all the complex coefficients in the predistortion table. This kind of constant value, and in particular the gain component, can be detected and eliminated given that, at the lowest transmitted power, the gain correction must normally be unitary (modulus of the first values in the predistortion table).
Some variants of the above methods can improve performance or simplify implementation, including:
the simplified (clipped) LMS technique, which reduces some complex multiplications for updating coefficients at the sampling rate to sign multiplications, without reducing the rate of convergence or the accuracy of linearization, and
two-dimensional predistortion to address wideband problems, such as nonlinearities that vary with the instantaneous frequency of the signal or the power of the samples previously amplified (short-term thermal memory effect).
In all cases, the algorithm is sensitive to shifting of the DC component, which must be eliminated. This task can be effected easily by a simple narrowband filter if a baseband ADC is used in the receiver.
In the prior art, the problem of correct alignment between the input signal and the measured signal has not been addressed, as a result of which the alignment is limited to xc2x1xc2xd sample and the measured signal difference error is reduced only by the ADC using the highest possible sampling rate and thus minimizing the shift error. At present, with approximately 80 MSPS and a signal bandwidth of 15 MHz, the limitation in the adaptive linearization gain is in the range from 12 dB to 15 dB.
Another method known in the art which can be envisaged if a good metric is available for measuring the time shift consists of delaying the source signal in a programmable manner to lock it onto the signal from the measurement receiver. A programmable filter is then used whose gain curve is flat and whose time-delay is a fraction of the sample time in the required band. This approach is very costly in terms of multipliers (for example Farrow filters, as described in the patent document WO 99/57806), because it operates at the sampling rate.
Given the above problems, an object of the invention is to provide a technique for time alignment of signals having a common origin and which, in one application, can dynamically calibrate and improve predistortion performance, implemented in digital form, in order to improve radio performance and the electrical efficiency of the radio amplifier.
The solution of the invention is both reliable and inexpensive to implement.
To be more precise, a first aspect of the invention provides a method of preparing signals to be compared to establish predistortion at the input of an amplifier, the signals comprising a signal before amplification and a signal after amplification by said amplifier,
which method is characterized in that the preparation includes performing time alignment between the signal before amplification and the signal after amplification, prior to using them to establish said predistortion.
The method advantageously comprises:
a coarse time alignment step, in which the signal before amplification is subjected to a time-delay comprising an integer number of first time units and in which the value of the time-delay that produces the delayed signal before amplification having the best time alignment with the signal after amplification is determined, and
a fine time alignment step, in which a delay or an advance value of a fraction of the first time unit to be applied to the sampled signal after amplification producing the best time alignment with the delayed signal before amplification is determined and in which that delay or advance value is applied to the signal after amplification,
the signals time aligned in this way being used to establish said predistortion.
The signal before amplification and the signal after amplification can be subjected to the first time alignment step in digital form.
The delayed signal before amplification having the best time alignment and/or the fraction of the first time unit may be determined by analyzing correlation points between the delayed signal before amplification and the signal after amplification, the delayed signal before amplification having the best time alignment being that which offers the highest correlation points.
The analysis of correlation points may then be applied to a plurality of separate delayed forms of the delayed signal before amplification subjected to the analysis in parallel, the delayed signal before amplification having the best time alignment being selected from the different forms of the delayed signal before amplification.
The analysis of correlation points may be based on the complex representation of at least one of the signals subject to the correlation, for example on the complex representation, i.e. the real and imaginary parts, of the delayed signal before amplification and on only the real representation of the signal after amplification.
In this implementation, the signal after amplification is digitized before the coarse time alignment step at a sampling frequency harmonically related to the sampling frequency of the signal before amplification.
A correlation curve may be defined for the delayed signal before amplification having the best time alignment with the aid of at least three correlation points, of which a first correlation point corresponds to the highest correlation value of the correlation points and the second and third correlation points are on respective opposite sides of the first correlation point, the alignment of the sampling times of the signals in the fine time alignment step being carried out by:
delaying the sampling time of the signal after amplification amplified by the amplifier relative to the delayed signal before amplification if the second correlation point has a lower correlation level than the third correlation point, and
advancing the sampling time of the signal after amplification amplified by the amplifier relative to the delayed signal before amplification if the second correlation point has a higher correlation level than the third correlation point,
or, conversely,
so as to converge towards a substantially equal correlation level between the second and third correlation points.
The correlation may be established taking into account only the signs of the delayed signal before amplification and the signal after amplification amplified by the amplifier and based on calculations of coincidence of signs between the two signals.
The signal after amplification is preferably digitized at a controlled phase sampling frequency, the fine time alignment step varying the sampling phase selectively to increase the time alignment of the delayed signal before amplification having the best time alignment and said signal after amplification amplified by the amplifier.
The first phase controlled sampling frequency may be produced at the output of a first phase-locked loop and the variation obtained by varying in an impulse fashion the reference frequency of that loop.
The reference frequency of the first phase-locked loop is advantageously produced by means of a second phase-locked loop, after division of its output frequency by a variable division number, and the variation of the reference frequency of the first phase-locked loop produced by changing the number to obtain said alignment of the phases of the delayed signal before amplification having the best time alignment and the signal after amplification.
The signal before amplification may be subjected to digital-to-analog conversion before it is fed to the input of the amplifier and the conversion process may be clocked by a sampling signal produced at the output of the second phase-locked loop.
A second aspect of the invention provides apparatus for preparing signals intended to be supplied to predistortion means operating at the input of an amplifier, the signals comprising a signal before amplification and a signal after amplification amplified by the amplifier,
which apparatus is characterized in that it includes means for time aligning the signal before amplification with the signal after amplification.
The optional aspects of the invention defined in the context of the method (first aspect) apply mutatis mutandis to the apparatus (second aspect).
A third aspect of the invention provides an amplifier system including an amplifier and predistortion means accepting input signals for comparison respectively before and after distortion by the amplifier, which amplifier system is characterized in that it includes time alignment apparatus according to the second aspect of the invention supplying the input signals for comparison.
The amplifier system can be used for linear power amplification in a wideband radio transmitter, for example in a band established on a frequency of the order of 10 MHz to 100 MHz, for example 60 MHz, and used to transmit multiple-carrier signals, for example for code division multiple access (CDMA) transmission for mobile telephones.